[..] |
ASTERISK-21002: Originate without Exten header does not work |
ASTERISK-21003: MOH keeps playing for the fist participant, if two participants connect at the same time to an empty conference |
ASTERISK-21004: Open Blockers for 1.8.21.0 |
ASTERISK-21005: Open Blockers for 11.3.0 |
ASTERISK-21006: unsupported host os "linux-gnueabihf" |
ASTERISK-21007: Remove Non Real time SIP Peers Automatically |
ASTERISK-21009: xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client |
ASTERISK-21011: Asterisk to Asterisk IAX2 trunk registration does not register |
ASTERISK-21012: Memory Leak on res_calendar (icalendar) |
ASTERISK-21013: Security Vulnerability: sip username disclosure |
ASTERISK-21014: logger.c Call_ID 'bound' or 'removed' DEBUG messages spammed during a feature code attended transfer |
ASTERISK-21016: One-way audio with JABBER correlates with 'res_xmpp.c: JABBER: socket read error' in debug |
ASTERISK-21019: PBX Cant Initialize (device_state_db - line 10492) |
ASTERISK-21020: Add support for legacy sip.conf configuration in res_sip's supported sorcery backends |
ASTERISK-21021: SQL script to create queue_log table in PostgreSQL |
ASTERISK-21024: Implement stasis-http GET /api/channels |
ASTERISK-21025: Implement stasis-http GET /api/channels/{channelId} |
ASTERISK-21026: Implement stasis-http POST /api/channels/{channelId}/answer |
ASTERISK-21027: Implement stasis-http DELETE /api/channels/{channelId} |
ASTERISK-21028: Implement stasis-http POST /api/channels/{channelId}/continue |
ASTERISK-21035: [patch] - features.conf in static realtime requires distinct cat_metric for each parking lot |
ASTERISK-21036: Jitter Buffer log file creation doesn't account for multiple slashes in DAHDI channel names |
ASTERISK-21037: skinny global vmexten and immed dial dont reset on module reload |
ASTERISK-21038: CLI: "core set debug channel" auto-complete returns "all", but not the names of available channels |
ASTERISK-21039: ODBC functions time out |
ASTERISK-21040: Deadlock involving chan_sip.c, pbx.c and autoservice.c, locking on chan and &conclock |
ASTERISK-21041: Asterisk crashes during a frame copy while receiving a fax |
ASTERISK-21042: [patch] - pbx_spool: callfile variables overriding/lost in __ast_request_and_dial() |
ASTERISK-21043: Motif/XMPP/Google Voice based calls keep ringing after re-establishing XMPP connection after socket error |
ASTERISK-21044: Call quality degrades after thousands of calls over a short period |
ASTERISK-21045: Session refresh reinvites an in progress T.38 dialog back to G.711 |
ASTERISK-21046: res_xmpp refcount issue |
ASTERISK-21047: asterisk crashes during an attended transfer SIP -> SIP -> DAHDI |
ASTERISK-21050: Asterisk crash during startup - issues with dlclose() return code checks and module loading registration |
ASTERISK-21051: Bridge API Enhancements: Refactor callers of ast_bridge_call to use Bridging API model |
ASTERISK-21052: Bridge API Enhancements: Implement threading model for bridge management thread |
ASTERISK-21054: Bridge API Enhancements: Add roles to the bridging model |
ASTERISK-21057: Bridge API Enhancements - add Stasis-Core events |
ASTERISK-21058: Bridge API Enhancements - rework Local channels/Local channel bridging |
ASTERISK-21059: Bridge API Enhancements - Refactor the Park family of applications |
ASTERISK-21061: Nortel I2004 unwanted autoanswer |
ASTERISK-21062: Pedantic should also be per extension directive |
ASTERISK-21063: Fix some issues with skinny callid |
ASTERISK-21064: Crash when handling ACK on dialog that has no channel |
ASTERISK-21065: Asterisk 11 IPv6 - FastAGI fail |
ASTERISK-21066: Respect Callerid ID presentation |
ASTERISK-21067: pointers of old channel_generator are not tidied up when a new channel_gernerator is activated |
ASTERISK-21068: Asterisk is freezing (since 1.8.18.0 to 1.8.20.1) when doing 'core show channels' AND receiving 'SIP register' |
ASTERISK-21069: xmpp distributed device states aggregation update fails |
ASTERISK-21070: DBdeltree throws spurious error under almost all cases |
ASTERISK-21071: [patch] Open channel for incoming call after RING; +CLIP responses from rfcomm; faster reporting of incoming calls |
ASTERISK-21072: Implement directmedia in chan_gulp |
ASTERISK-21074: Implement NAT settings in chan_gulp |
ASTERISK-21076: Implement non-session based messaging support (RFC 3428) |
ASTERISK-21077: Add support for video in chan_gulp |
ASTERISK-21080: Redial button does not work properly |
ASTERISK-21081: New SIP Channel Driver - Registrar - Part One |
ASTERISK-21082: New SIP Channel Driver - Registrar - Part Two |
ASTERISK-21083: New SIP Channel Driver - Registrar - Part Three |
ASTERISK-21084: New SIP Channel Driver - Path Support |
ASTERISK-21089: New SIP Channel Driver - Test Plan for Basic Calls |
ASTERISK-21091: Add 0x144 skinny support |
ASTERISK-21094: MixMonitorMute mutes through stream if already slinear (e.g. Originate) |
ASTERISK-21095: More called details fixup |
ASTERISK-21096: Complete channel snapshot work for Stasis Core |
ASTERISK-21097: Stasis Core - Refactor MWI support |
ASTERISK-21098: Asterisk 1.8.12.0 core dumps |
ASTERISK-21099: Reload makes dahdi not work |
ASTERISK-21101: Stasis Core - Refactor Device State support |
ASTERISK-21102: Stasis Core - Refactor Presence State support |
ASTERISK-21103: Stasis Core - Refactor the other event types onto the Stasis Core message bus |
ASTERISK-21108: If chan_motif fails to load, Asterisk still thinks it's loaded |
ASTERISK-21111: segfault Asterisk 1.6.0.9 |
ASTERISK-21113: app_dial.c does not honor 'c' flag when calling party hangs up |
ASTERISK-21117: Bad interpretation of the file chan_dahdi.conf when using open r2 parameters |
ASTERISK-21119: Asterisk system locks up with chan_unistim |
ASTERISK-21120: Unable to properly hang up calls when second line rings |
ASTERISK-21122: Documentation on the various methods of changing verbosity can be confusing |
ASTERISK-21123: Compilation error: pjproject/build.mak: version.mak: No such file or directory |
ASTERISK-21124: cdr_adaptative_odbc does not populate cdr information |
ASTERISK-21125: Asterisk 11 needs libuuid in configure script due to pjproject |
ASTERISK-21127: Empty custom CDR value |
ASTERISK-21128: Locking inversion when attempting to set caller ID while holding iaxsl lock causes deadlock |
ASTERISK-21129: Lock on do_monitor |
ASTERISK-21130: sip_pvt.dsp incorrect manipulation related to inband dtmfmode and faxdetect in SIPDtmfMode() app. and enable_dsp_detect() |
ASTERISK-21131: [patch] - Asterisk creates SDP with (peer) unsupported audio codec |
ASTERISK-21133: SIP/TDM interworking, and RTP on CALL PROCEEDING |
ASTERISK-21134: On Asterisk 11.2.1 can't load chan_dahdi.so |
ASTERISK-21135: Asterisk 1.8 no longer sends unsolicited message-summary (NOTIFY) after realtime SIP peer registers |
ASTERISK-21139: Asterisk 11 Seg Faults on READ after ConfBridge KICK |
ASTERISK-21141: RPID not parsed correctly if display-name is *(token LWS) |
ASTERISK-21142: Page app can't find ConfBridge |
ASTERISK-21144: One way audio after channels are AMI Bridged out of a ConfBridge that has jitterbuffer=yes |
ASTERISK-21145: asterisk doesn't continue to the next priority after a soft hangup |
ASTERISK-21146: Semi attended (blonde) transfer causes queue member not to respect wrapuptime |
ASTERISK-21148: [patch] - Asterisk use '(null)' in 'via' header and 'call-id' header when relaying SIP MESSAGE |
ASTERISK-21149: detailed hangup cause for ${REASON} variable in call files |
ASTERISK-21150: UDPTL Error Correction Scheme Negotiation Issue, Asterisk copies the Error Correction Scheme from T38 offer from remote peer even if UDPTL error correction scheme is set to NONE for that peer in sip.conf |
ASTERISK-21151: 'Squelching' early media in DAHDI (sig_pri) |
ASTERISK-21152: if pressed * the user menu is not working |
ASTERISK-21153: Getting duplicate entry in CDR entry for same call |
ASTERISK-21155: CallerID on Indian PSTN is not working. |
ASTERISK-21156: Asterisk crashes with XMPP\Google Voice config where username is missing hostname portion |
ASTERISK-21157: Asterisk 1.8.20.0 Crash when unloading chan_dahdi |
ASTERISK-21158: Video enabled peers will send a video stream when calling a voice only peer. |
ASTERISK-21160: In an XMPP distributed device state configuration, setting pubsub_node to a value without the pubsub prefix can cause a dialog loop leading to high CPU usage and crashiness |
ASTERISK-21162: Deadlock in cdr.c: cdr_batch_lock vs cdr_pending_lock |
ASTERISK-21163: pjproject raises an assert failure when creating TURN socket while adding ICE candidates to RTP session |
ASTERISK-21164: Need clarification on distributed device state behavior and whether this behavior is a possible regression |
ASTERISK-21168: asterisk logger stops logging VERBOSE and NOTICE messages after some time. |
ASTERISK-21170: DTMF timestamp issue |
ASTERISK-21172: One way audio when external Call forwarded to queue member |
ASTERISK-21173: [patch] example sippeers sql hasn't been adapted for ipv6 and causes chan_sip to generate a warning message |
ASTERISK-21174: Asterisk 11 auto-pause problem |
ASTERISK-21176: Call files on OS X, using KQueue, do not get processed (load 100%) |
ASTERISK-21177: [patch] Issues with skinny callinfo during fwd |
ASTERISK-21178: Improve documentation for manager command Getvar, Setvar |
ASTERISK-21180: Implement channel state events for Stasis HTTP |
ASTERISK-21182: Create documentation/specification for expected Stasis HTTP events |
ASTERISK-21184: chan_gulp: Add support for multiple media streams/types |
ASTERISK-21186: chan_gulp - Implement media negotiation rules |
ASTERISK-21190: chan_mgcp crash on chunked m= sdp line |
ASTERISK-21191: [patch] - VoiceMailPlayMsg doesn't work with ODBC |
ASTERISK-21193: IAX/2 fails to destroy channel on max retransmits exceeded |
ASTERISK-21194: chan_sip can fail to find a peer during reload |
ASTERISK-21195: Transcoding makes bad choice in high-rate translations |
ASTERISK-21196: Refactor CDRs onto Stasis-Core to handle changes in bridging behavior |
ASTERISK-21199: Implement outbound authentication handling support |
ASTERISK-21201: [patch] In Manager Interface, SIP registry event does not show username on Status: Registered |
ASTERISK-21202: Asterisk SIP message (SMS) stops working |
ASTERISK-21203: res_xmpp socket error: takes upto 19 minutes to restore xmpp socket connection to google |
ASTERISK-21204: Asterisk increments the session version in 2xx message even if a '183 Session in Progress' with SDP has already been sent in response to initial INVITE. |
ASTERISK-21205: [patch] dundi_read_result crash due to negative number |
ASTERISK-21206: Crashes contstantly in chan_motif |
ASTERISK-21207: [patch] - Deadlock on fax extension calling ast_async_goto() with locked channel |
ASTERISK-21208: 'sip set options {on|off}' command to explicitly enable 200 OK responses to OPTIONS |
ASTERISK-21209: crash in res_clialiases on reload |
ASTERISK-21210: BRI locks up |
ASTERISK-21211: chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault |
ASTERISK-21215: Unexpected Behavior in Adaptive CDR when U() used on Dial |
ASTERISK-21216: Skinny voicemail indication issues |
ASTERISK-21222: Behavior of 'logger set level' With Respect To Entries in logger.conf |
ASTERISK-21223: Asterisk no longer responds to SIP REGISTER's that don't contain an Authorization |
ASTERISK-21224: [patch] Skinny groupPickup issues |
ASTERISK-21225: [patch] Setting nat=force_rport in [general] sip.conf will never work |
ASTERISK-21226: one way audio after call was on hold |
ASTERISK-21228: Deadlock in pbx_find_extension when attempting an autoservice stop due to holding the context lock |
ASTERISK-21231: When outboundpoxy is set, asterisk should not attempt to resolve DNS |
ASTERISK-21232: Asterisk sends AUDIO REINVITE when session timer expires in T38 call |
ASTERISK-21233: [patch] Downgrade missing speeddials to a template debug and remove unsupported message for 7937 |
ASTERISK-21234: Deadlock when using two Local channels & fax gateway (local_queryoption) |
ASTERISK-21236: 11.3 release |
ASTERISK-21237: detect PRI_EVENT_NOTIFY(16) instead of PRI_EVENT_HANGUP_REQ(15) |
ASTERISK-21241: When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored |
ASTERISK-21242: Segfault when T.38 re-invite retransmission receives 200 OK |
ASTERISK-21243: [patch] Backport Appropiate NAT Setting Cleanups To 1.8 |
ASTERISK-21244: Crash in libsrtp when attempting to unprotect RTCP packet |
ASTERISK-21245: Application Dial - Option g ignored with option A if the called part hangs up during the announce. |
ASTERISK-21246: [patch] use of rtpkeepalive uses CN packet with marker bit set, plus a ULAW payload instead of CN |
ASTERISK-21248: CALLERID(dnid-num-plan) does not get any value set. |
ASTERISK-21250: AddQueueMember does not write membername to the queue log |
ASTERISK-21253: Create a test realtime backend suitable for driving Asterisk Test Suite tests |
ASTERISK-21255: Create a sorcery wizard for the AstDB |
ASTERISK-21257: Implement inbound/outbound Caller ID handling |
ASTERISK-21258: Implement mid-call connected line support for chan_gulp |
ASTERISK-21259: Build a pub/sub architecture for the new SIP channel driver |
ASTERISK-21260: Add MWI support to the new SIP channel driver |
ASTERISK-21261: Add DTMF Info support |
ASTERISK-21267: Add stasis-http configuration |
ASTERISK-21270: Bridge API Enhancements - add subclassing ability to ast_bridge |
ASTERISK-21271: Bridge API Enhancements - subclass ConfBridge with its own Virtual Method table |
ASTERISK-21272: Bridge API Enhancements - subclass Parking with its own Virtual Method table |
ASTERISK-21277: stasis-http authentication |
ASTERISK-21278: stasis-http Cross-Origin configuration |
ASTERISK-21279: Allow WebSocket connections on URL's other than /ws |
ASTERISK-21280: Basic configuration for stasis-core |
ASTERISK-21281: stasis-http: Create Confluence swagger-codegen templates |
ASTERISK-21282: Add DTMF events to the stasis-http WebSocket |
ASTERISK-21283: Implement stasis-http POST /api/channels/{channelId}/play |
ASTERISK-21292: Add callfwd_noanswer to skinny |
ASTERISK-21293: Script to calculate number of each character sequence occurrence specified as an argument |
ASTERISK-21294: Calling StopMixMonitor on a channel w/o MixMonitor running returns -1 |
ASTERISK-21295: Sip registration fails, wrong parsing when secret has parentheses symbol |
ASTERISK-21296: SIP module not responding any more |
ASTERISK-21297: Segmentation fault on hangup in in ast_bridged_channel |
ASTERISK-21298: Confbridge recording fails - deadlock |
ASTERISK-21299: Asterisk send wrong codec order in the leg B of the call |
ASTERISK-21300: Asterisk is sending wrong codec order in the leg B of the call |
ASTERISK-21301: ERROR and failure to resolve socket address due to whitespace after port number in SIP Via header |
ASTERISK-21302: [patch] app_voicemail crashes on config error and there are some potential memory leaks |
ASTERISK-21303: qualifygap SIP general setting appears broken |
ASTERISK-21304: [patch] AGI AsyncAGI event returns AGI command arguments |
ASTERISK-21305: Segfault when hanging up channels active in MeetMe with recording |
ASTERISK-21306: set FEATURE(parkingtime) is not inherited by child channels |
ASTERISK-21310: __sip_xmit fails with interrupted system call |
ASTERISK-21311: CLI command 'module load' attempts to free unallocated memory on tab completion |
ASTERISK-21314: Sip channel is deadlocked |
ASTERISK-21315: Asterisk Realtime using res_config_mysql not recognizing port field for SIP peers |
ASTERISK-21316: Segfault on ast_channel_tech(chan)->send_digit_begin |
ASTERISK-21318: AMI events for ConfBridge Mute/Unmute and Record start/stop |
ASTERISK-21320: fails to parse irregular version strings (such as ~dfsg) |
ASTERISK-21321: Skinny softkey endcall when transferring should not blind xfer |
ASTERISK-21322: fails to copy relative symlinks from the tree |
ASTERISK-21323: Asterisk 11 svn branch and srtp - white noise only |
ASTERISK-21324: [patch] Per-user option 'allowmultiplelogin' in manager |
ASTERISK-21325: astdb2sqlite is very slow when running the 32-bit version on 64 system |
ASTERISK-21327: Add transfer softkey when transferor chan ringing |
ASTERISK-21328: ISDN PRI Release Delay |
ASTERISK-21329: chan_alsa: patch for crash when audio device in unexpected state |
ASTERISK-21330: XML documentation generation fails on fresh checkouts |
ASTERISK-21331: Bridge API Enhancements - add bridging unique identifier, bridge container, and basic CLI commands |
ASTERISK-21332: Bridge API Enhancements - create the Basic Bridge subclass |
ASTERISK-21333: Bridge API Enhancements - refactor all uses of a jitter buffer to use func_jitterbuffer |
ASTERISK-21334: Bridge API Enhancements - hide masquerades |
ASTERISK-21335: Bridge API Enhancements - add externally initiated blind transfers |
ASTERISK-21336: Bridge API Enhancements - add externally initiated attended transfers |
ASTERISK-21337: Bridge API Enhancements - add stasis core messages for blind/attended transfers |
ASTERISK-21338: Bridge API Enhancements - Refactor the Dial API as a bridge mixing technology |
ASTERISK-21339: Bridge API Enhancements - add CCSS, Connected Line, Pre-Dial to the Dial API |
ASTERISK-21352: Bridge API Enhancements - refactor ParkAndAnnounce application to use the new parking bridge |
ASTERISK-21353: Bridge API Enhancements - add features to parking |
ASTERISK-21354: Bridge API Enhancements - perform basic parking pickup |
ASTERISK-21356: Segfault during bridge channel proxy inspection in a masquerade caused by an AMI Redirect of two channels |
ASTERISK-21359: Refactor AMI DTMF events onto Stasis-Core |
ASTERISK-21366: Transfer settings to compile asterisk menuselect.makeopts copying from old installation |
ASTERISK-21367: Executing StopMixMonitor on channel is not MixMonitored hangs up this channel |
ASTERISK-21368: Add Manager Events for SIP Registry status changing |
ASTERISK-21369: Need to INVITE to peer with other domain without peer domain addition |
ASTERISK-21370: Call gets dropped transferring to external destination |
ASTERISK-21373: In proxy NAT traversal situation the far end requires Asterisk to send RTP first |
ASTERISK-21374: [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX |
ASTERISK-21377: After sip reload 80% peers is UNREACHEBLE |
ASTERISK-21378: chan_sip completely blocks on DNS lookups |
ASTERISK-21382: make (at least some of) the payload numbers used by dynamic payload types configurable |
ASTERISK-21383: STUN Binding Requests Not Being Sent Back from Asterisk to Chrome |
ASTERISK-21384: Unique ID Call Count Increasing By 2 |
ASTERISK-21385: SIP Channel Lock |
ASTERISK-21386: SIP Channel Locks |
ASTERISK-21387: Asterisk unresponsive while waiting for lock MUTEX 6239 __ast_pbx_run c |
ASTERISK-21389: res_timing_pthread fails to return from write, causing timer dependent operations to block indefinitely |
ASTERISK-21390: New applications for app_stack.c: GosubEntry and StackPopGoto |
ASTERISK-21391: Asterisk fails to load chan_sip.so after compiling with "#define REF_DEBUG 1" in chan_sip.c |
ASTERISK-21394: [patch] - Fundamental changes to CDR within single asterisk family (1.8) during externally initiated blind transfers with an h extension present |
ASTERISK-21395: High CPU use due to invalid time limit passed to kevent() in pbx_spool.c:scanthread |
ASTERISK-21397: [patch] manager crash on unloading app_queue |
ASTERISK-21398: [patch] chan_iax2.c:7998 authenticate_verify: requested inkey 'my_oth' for RSA authentication does not exist |
ASTERISK-21399: RTP Multicast of L16 (type 10): Asterisk and wireshark disagree |
ASTERISK-21400: Websocket Call Ends in Match Not Found |
ASTERISK-21401: [patch] codec_resample cannot be unloaded |
ASTERISK-21402: [patch] Unloading app_queue can cause AMI to segfault |
ASTERISK-21406: [patch] chan_sip deadlock on monlock between unload_module and do_monitor |
ASTERISK-21407: [patch] features_shutdown doesn't finish cleanup |
ASTERISK-21409: [patch] - Race condition with IAX2 transfer, 2 releases happen on same call legs. locks up with many threads blocked by iax2_destroy_helper |
ASTERISK-21410: Park() application never returns in some cases |
ASTERISK-21411: Far end Re-invites Asterisk without SDP (ACKs with SDP) - Asterisk does not modify port of RTP stream - Ignores ACK due to no SDP version change |
ASTERISK-21412: [patch] config.c/config_text_file_load() leaks globbuf |
ASTERISK-21413: app_voicemail sound file for forwarding messages can be misleading |
ASTERISK-21414: Recording with MixMonitor consumes more G.729 licenses than expected on a single call |
ASTERISK-21415: Asterisk Segmentation Fault when reloading module a few times in a short delay |
ASTERISK-21416: Implement SDES-SRTP support in chan_gulp |
ASTERISK-21419: Implement DTLS-SRTP support in chan_gulp |
ASTERISK-21421: API Improvements: build out the concept of an endpoint in Stasis-Core |
ASTERISK-21422: Asterisk Test Suite - rework our CEL testing module to be reliable |
ASTERISK-21424: Implement chan_gulp tests - off nominal incoming call paths |
ASTERISK-21426: New SIP Channel Driver - Call Forwarding |
ASTERISK-21429: Distributed Device State using JABBER/XMPP not working since Secuity Advisory AST-2012-015 |
ASTERISK-21430: [patch] Call ID missing when logging through syslog |
ASTERISK-21432: Video isn't negotiated when endpoint switches to video on an established SIP to SIP call |
ASTERISK-21433: Add analogous support for 'alwaysauthreject' to chan_gulp and top level security settings |
ASTERISK-21434: Add anonymous access support to chan_gulp |
ASTERISK-21435: Add redirecting information support to chan_gulp |
ASTERISK-21436: Add CLI/AMI initiated NOTIFY requests (sip_notify support) |
ASTERISK-21441: New SIP Channel Driver: Create an API on top of the pub/sub framework for extension state notifications |
ASTERISK-21442: New SIP Channel Driver - Create an extension state provider for RFC 3863 |
ASTERISK-21443: New SIP Channel Driver - Create a state provider for dialog-info+xml |
ASTERISK-21447: Asterisk crashes while connecting to TCP peers |
ASTERISK-21448: New SIP Channel Driver - basic fax support |
ASTERISK-21450: Allow pluggable modules to be executed against particular Asterisk Versions |
ASTERISK-21452: New SIP Channel Driver - Create Event State Compistor resource module and implement Publish API |
ASTERISK-21453: New SIP Channel Driver - Implement CCSS |
ASTERISK-21456: New SIP Channel Driver - add basic REFER support and SIP blind transfers |
ASTERISK-21457: New SIP Channel Driver - enhance basic REFER support to handle SIP attended transfers |
ASTERISK-21460: New SIP Channel Driver - create a SIP Security Event module suitable for consumption in the new SIP stack |
ASTERISK-21462: Stasis Core - Refactor random AMI events |
ASTERISK-21464: with directrtpsetup some payload type identifiers from A party's INVITE are not copied to the INVITE for B party |
ASTERISK-21465: wrong routing of ACK request following a 200OK (Record-Route header not taken into account) |
ASTERISK-21466: [patch] [crash] command (sip show peers) crashes Asterisk with ~3500 registered peers |
ASTERISK-21467: Stasis Core - Refactor MeetMe Events |
ASTERISK-21468: Stasis Core - Refactor ConfBridge Events |
ASTERISK-21469: Stasis Core - Refactor Queue Events |
ASTERISK-21470: Stasis Core - Refactor AGI Events |
ASTERISK-21471: Stasis Core - Refactor RTP/RTCP Events |
ASTERISK-21472: Stasis Core - Refactor AOC Events |
ASTERISK-21473: Stasis Core - Refactor CCSS events to Stasis-Core |
ASTERISK-21474: Stasis Core - Refactor AddOn Channels |
ASTERISK-21475: CallerID information doesn't persist after a Channel Redirect on a H323 leg (works with other channel technologies) |
ASTERISK-21476: Stasis Core - Refactor extraneous channel events |
ASTERISK-21486: Call answered by a dynamic agent and then SIP transferred to an external number is not written to CDR |
ASTERISK-21487: Stasis Core - Refactor Hold event from chan_sip/chan_iax2/sig_pri to channel core |
ASTERISK-21488: Stasis Core - Refactor Registry events from chan_iax2/chan_sip |
ASTERISK-21489: Stasis Core - Refactor PeerStatus events |
ASTERISK-21492: RTP packetization negotiated at ptime=30 for first leg; ptime=20 for second leg results in deltas of 20/40ms and 30/0ms |
ASTERISK-21494: AMI 1.4 Improvements - Add a field to all AMI events that conveys the system name |
ASTERISK-21495: 302 Moved Temporarily CDR Incorrect |
ASTERISK-21496: Stasis Core - Add the Transfer bridging message and corresponding AMI event |
ASTERISK-21499: New SIP Channel Driver - add/finish up SIP Qualify Support |
ASTERISK-21500: New SIP Channel Driver - Add provisional keep alives |
ASTERISK-21501: New SIP Channel Driver - add custom INFO support for one touch recording |
ASTERISK-21502: New SIP Channel Driver - add Advice of Charge support |
ASTERISK-21503: New SIP Channel Driver - integrate stasis endpoints |
ASTERISK-21504: New SIP Channel Driver - add SIP History tracking |
ASTERISK-21505: New SIP Channel Driver - add call pickup group configuration options |
ASTERISK-21506: New SIP Channel Driver - add a variety of customization configuration parameters |
ASTERISK-21507: New SIP Channel Driver - add progressinband |
ASTERISK-21517: API Improvements: refactor app_queue to listen for a Transfer stasis message and update the Queue Log appropriately |
ASTERISK-21518: Bridge API Enhancements - refactor chan_iax2 to perform blind transfers using the new bridging framework |
ASTERISK-21519: Bridge API Enhancements - implement blind transfers in chan_sip |
ASTERISK-21520: Bridge API Enhancements - implement attended transfers in chan_sip |
ASTERISK-21522: [patch] DTMF end is not always processed, causes one-way audio |
ASTERISK-21523: Bridge API Enhancements - refactor sig_pri_attempt_transfer to use Bridging Framework |
ASTERISK-21524: Bridge API Enhancements - refactor chan_misdn's misdn_attempt_transfer |
ASTERISK-21525: Bridge API Enhancements - refactor chan_mgcp attempt_transfer |
ASTERISK-21526: Bridge API Enhancements - refactor chan_skinny skinny_transfer |
ASTERISK-21527: Bridge API Enhancements - refactor chan_unistim attempt_transfer |
ASTERISK-21542: Bridge API Enhancements - get DTMF attended transfers feature complete - configuration support |
ASTERISK-21543: Bridge API Enhancements - get DTMF attended transfers feature complete - add attended transfer monitoring |
ASTERISK-21544: Bridge API Enhancements - get call pickup working |
ASTERISK-21549: AMI 1.4 Improvements - refactor ast_pbx_outgoing_* to use the dial API; add Originate AMI Events |
ASTERISK-21550: AMI 1.4 Improvements - Add Dial Begin/End messages to FollowMe |
ASTERISK-21551: AMI 1.4 Improvements - Add Dial Begin/End messages to Queue |
ASTERISK-21552: AMI 1.4 Improvements - Refactor AMI/CLI channel inspection actions to use the stasis channel cache |
ASTERISK-21553: Bridge API Enhancements - add one touch recording |
ASTERISK-21554: Bridge API Enhancement - do something about chan_agent |
ASTERISK-21555: Bridge API Enhancements - implement channel variables in the bridging core |
ASTERISK-21563: API Enhancements - CEL refactoring - channel state |
ASTERISK-21564: API Enhancements - CEL refactoring - bridge state |
ASTERISK-21565: API Enhancements - CEL refactoring - transfers |
ASTERISK-21566: API Enhancements - CEL refactoring - cleanup |
ASTERISK-21567: API Enhancements - CEL refactoring - Documentation |
ASTERISK-21573: CallCompletionRequest() not available with call forwarding disabled on busy extension |
ASTERISK-21574: Queue is sending multiple calls to the available agents at once when autofill is enabled |
ASTERISK-21575: Asterisk REST API - Implement GET /asterisk/info call |
ASTERISK-21576: Asterisk REST API - Implement /recordings |
ASTERISK-21577: Asterisk REST API - Update the /recording template |
ASTERISK-21578: Asterisk REST API - Create the sounds resource template |
ASTERISK-21579: Asterisk REST API - Create the playback template |
ASTERISK-21580: Asterisk REST API - Update bridge/channel templates for playback and record operations |
ASTERISK-21581: Asterisk REST API - Implement GET /recording/{id} |
ASTERISK-21582: Asterisk REST API - Implement DELETE /recording/{id} |
ASTERISK-21583: Asterisk REST API - Implement POST /recording/{id}/rename |
ASTERISK-21584: Asterisk REST API - Implement GET /sounds |
ASTERISK-21585: Asterisk REST API - Implement GET /sound/{id} |
ASTERISK-21586: Asterisk REST API - implement GET /playback/{id} |
ASTERISK-21587: Asterisk REST API - Implement POST /playback/{id}/control |
ASTERISK-21588: Asterisk REST API - Implement POST /playback/{id}/restart |
ASTERISK-21589: Asterisk REST API - Implement POST /playback/{id}/pause |
ASTERISK-21590: Asterisk REST API - Implement POST /playback/{id}/rewind |
ASTERISK-21591: Asterisk REST API - Implement POST /playback/{id}/fastforward |
ASTERISK-21592: Asterisk REST API - Implement POST /bridge/{id}/play |
ASTERISK-21593: Asterisk REST API - Implement POST /bridge/{id}/record |
ASTERISK-21594: Asterisk REST API - Implement POST /channel/{id}/record |
ASTERISK-21615: Asterisk REST API - Implement GET /endpoints |
ASTERISK-21616: Asterisk REST API - Implement GET /endpoint/{id} |
ASTERISK-21617: Asterisk REST API - Implement POST /channels to an endpoint |
ASTERISK-21618: Asterisk REST API - Implement POST /channels/{id}/mute and /channels/{id}/unmute |
ASTERISK-21619: Asterisk REST API - Implement POST /channel/{id}/hold and /channel/{id}/unhold |
ASTERISK-21620: Asterisk REST API - Implement POST /channel/{id}/dial |
ASTERISK-21621: Asterisk REST API - Implement GET /bridges |
ASTERISK-21622: Asterisk REST API - Implement GET /bridge/{id} |
ASTERISK-21623: Asterisk REST API - Implement DELETE /bridge/{id} |
ASTERISK-21624: Asterisk REST API - Implement POST /bridges |
ASTERISK-21625: Asterisk REST API - Implement POST /bridge/{id}/addChannel |
ASTERISK-21626: Asterisk REST API - Implement POST /bridge/{id}/removeChannel |
ASTERISK-21639: Segfault in app_confbridge while stress testing |
ASTERISK-21640: Bridge API Enhancements - work through a channel being removed from a bridge by an external party |
ASTERISK-21641: Bridge API Enhancements - get Park AMI action working again |
ASTERISK-21642: Bridge API Enhancements - add hints back to the parking slots |
ASTERISK-21643: Bridge API Enhancements - add the default parking lot |
ASTERISK-21644: Bridge API Enhancements - add dynamic parking lots |
ASTERISK-21645: Bridge API Enhancements - add parking dialplan generation |
ASTERISK-21654: DNS SRV lookup doesn't bother with family (ipv4, ipv6) |
ASTERISK-21657: asterisk locks up after running traffic |
ASTERISK-21658: Asterisk REST API - Implement POST /channels to a dialplan context/extension/priority |
ASTERISK-21660: Queue does not respect agent status - deliver a call even agent is busy or unavailable |
ASTERISK-21661: SMS delvery to VoIP mobile |
ASTERISK-21662: Res_odbc keeps losing connection to MySQL |
ASTERISK-21663: [patch] Realtime TCP endpoints lose registration after "sip reload" & "core reload" |
ASTERISK-21664: Asterisk terminates calls if Session-Expires isn't present on INVITE |
ASTERISK-21665: 11.X Crash on debian/sparc with SIGBUS, Bus Error |
ASTERISK-21666: patch to implement match_auth_username option(sip.conf) for SIP REGISTER |
ASTERISK-21667: No AMI events output until Asterisk receives an AMI command |
ASTERISK-21668: Basic res_sip XML documentation |
ASTERISK-21669: Fix dependencies on res_sip files |
ASTERISK-21670: Coding style within chan_gulp |
ASTERISK-21671: Asterisk realtime/ SIP status |
ASTERISK-21672: Early media not properly handled on outbound TCP trunk |
ASTERISK-21673: Asterisk conversion of 183 without SDP to 180 breaks interoperability with Microsoft Lync |
ASTERISK-21675: Asterisk forgot G729 license |
ASTERISK-21676: (CALLERPRES()=prohib) not honoured over ISDN |
ASTERISK-21677: NOTIFYs for BLF start queuing up and fail to be sent out |
ASTERISK-21678: IPv6-configured sip channel transmits to (null) for IPv4 register= hosts |
ASTERISK-21683: Asterisk 1.8.21.0 Blind Transfer To Parking For An Inbound Call Fails And Leaves Call In Limbo State |
ASTERISK-21688: CDR record cannot be modified in 'h' extension when 'g' option is used in Dial application |
ASTERISK-21689: AMI bridge continues to try and bridge even after reporting "Channel2" does not exist; and fails resulting in hangup of Channel1 |
ASTERISK-21690: Asterisk sends SIP 481 after REFER |
ASTERISK-21691: bridge cmd does not hangup if the bridged leg hangedup |
ASTERISK-21693: Use of possibly uninitialized value in ast_channel_hangupcause_hash_set |
ASTERISK-21694: Peer with outbound proxy is resolved in DNS |
ASTERISK-21695: Crash Asterisk |
ASTERISK-21696: Assertion error results in crash in pjproject's ICE worker thread |
ASTERISK-21697: Bridge API Enhancements - handle Local Channel Optimization in CDRs |
ASTERISK-21698: Bridge API Enhancements - handle Attended Transfers in CDRs |
ASTERISK-21699: Bridge API Enhancements - handle Call Pickup in CDRs |
ASTERISK-21703: New SIP Channel Driver - write a registration test plan on the wiki |
ASTERISK-21708: Bridge API Enhancements - write a test plan for Queues |
ASTERISK-21710: New SIP Channel Driver - implement the promiscredir option in chan_gulp |
ASTERISK-21711: Stasis API - Incorporate the bridging framework into res_stasis app |
ASTERISK-21713: Bridge API Enhancements - Create a media channel for the bridging API |
ASTERISK-21716: [patch] logger thread sometimes exits with messages still queued |
ASTERISK-21717: [patch] - Documentation for PASSTHRU function is unclear |
ASTERISK-21718: [patch] pbx_dundi leaks ast_io_add |
ASTERISK-21719: [patch] res_srtp doesn't cleanup srtp library |
ASTERISK-21720: Asterisk 11 cannot compile with multiple definitions. Possible libasteriskssl + openssl issue. |
ASTERISK-21721: SIP Failed to parse multiple Supported: headers |
ASTERISK-21722: chan motif behaves wrong |
ASTERISK-21723: [patch] pbx cleanup is incomplete |
ASTERISK-21724: [patch] __ast_rwlock_destroy can segfault with DEBUG_THREADS |
ASTERISK-21725: Asterisk 11 attempts IPv6 (with an insane address) when talking to an IPv4-only endpoint |
ASTERISK-21726: Asterisk does not properly parse multiple allow: headers |
ASTERISK-21737: [patch] - Crash during transfer from DAHDI/SIP to SIP/SIP in ast_format_cap_append called from remote bridge loop |
ASTERISK-21738: [patch] Segfault On Realtime Queue Members Processing |
ASTERISK-21741: [patch] - Improved Caller ID Diagnostics and Processing for FXO Channels |
ASTERISK-21742: SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher. |
ASTERISK-21743: [patch] - Core show Locks, Include Asterisk version. |
ASTERISK-21744: [patch] - fix lower bound check with -ve integer conversion from a float |
ASTERISK-21751: Asterisk crashes with segmentation fault when trying to do a pickup with INVITE with Replaces |
ASTERISK-21752: Asterisk peers with host=<dnsname> do not accept calls from all hosts in dnsname's multiple host SRV record set |
ASTERISK-21753: Seg Fault while attempting to queue AST_CONTROL_SRCCHANGE on a NULl channel when handling an incoming SIP ACK over TCP |
ASTERISK-21754: Bridge API Enhancements - write tests for the new Bridge AMI actions/events |
ASTERISK-21756: assert() when using dtmfmode=none |
ASTERISK-21757: segfault on asterisk startup: motif iksemel |
ASTERISK-21758: chan_gtalk and res_xmpp not compiling with openssl-devel installed if iksemel-devel is not |
ASTERISK-21760: Asterisk autoconf script does not check for pkg-config as a dependency |
ASTERISK-21761: sip call stuck, crash |
ASTERISK-21762: IAX2 call problem |
ASTERISK-21763: asterisk -r Bus Error on Debian/sparc |
ASTERISK-21765: [patch] - FILE function's length argument counts from beginning of file rather than the offset |
ASTERISK-21768: LICENSE file missing from Asterisk Extra sounds (French) |
ASTERISK-21772: Redundant if statement in dns.c |
ASTERISK-21773: Asterisk 1.8.22.0 Open Blockers |
ASTERISK-21774: Asterisk 11.4.0 Open Blockers |
ASTERISK-21775: FPE during MOH playback |
ASTERISK-21777: Asterisk tries to transcode video instead of audio |
ASTERISK-21778: astobj2.c:115 INTERNAL_OBJ: user_data is NULL followed by Segmentation fault on cancelled divert |
ASTERISK-21779: Manager closes connection when a SendText action is requested during hangup |
ASTERISK-21780: Add missing documentation for new config option |
ASTERISK-21781: on reload app_queue should check/prune queue members against associated devices in configuration |
ASTERISK-21782: Delayed audio to agent when answering a queue call |
ASTERISK-21785: __ao2_ref_debug() logs to /tmp/refs when REF_DEBUG is not defined |
ASTERISK-21786: Segfault in MyODBC MySQL connector during reconnect attempt when connection is lost |
ASTERISK-21787: No IAX2 communication either user/peer or friend accounts |
ASTERISK-21788: What variable stores caller queue position |
ASTERISK-21789: ast_http_get_cookies() fails in the presence of RFC2965 Cookie2 header |
ASTERISK-21792: chan_sip.c: Autodestruct on dialog X with owner X in place (Method: BYE). Rescheduling destruction for X |
ASTERISK-21793: Segmentation fault when dealing with Agent channels |
ASTERISK-21794: CLI command 'realtime update2' syntax failure when using according to usage help |
ASTERISK-21795: failed compilation - dns.c references res_nsearch which is not available on uclibc |
ASTERISK-21797: FILE function reads inconsistently |
ASTERISK-21799: [patch] Dropouts/distortion in MixMonitor recording when recording RTP with ptime of 60ms |
ASTERISK-21800: ooh323 channels stuck if no gatekeer or ooh323 reload |
ASTERISK-21802: (un)muting a ConfBridge user via *CLI doesn't generate AMI events |
ASTERISK-21803: transcoding from silk to g711 constantly print the message "lintosilk_frameout: encoding XXX samples" |
ASTERISK-21804: CLI command 'reload' has inconsistent output written to console when operating with a DEBUG level greater than 0 |
ASTERISK-21807: Wrong reference and missing values for "Hangup Hause Code Mappings" |
ASTERISK-21809: [patch] sip_pvt members novideo and notext are being reset to TRUE every time SDP is processed |
ASTERISK-21811: cdr_odbc "CDR direct execute failed" Not working with MySQL Master Server |
ASTERISK-21812: Whitespace not escaped correctly on AGI() |
ASTERISK-21814: "/" in TOUCH_MIXMONITOR is replaced by "-" |
ASTERISK-21815: SipRemoveHeader does not remove previously added Alert-Info Header |
ASTERISK-21816: [patch] OpenBSD fix for UUID |
ASTERISK-21817: Stasis-HTTP: Implement Stasis message_type formatting functions |
ASTERISK-21818: Stasis-HTTP: Write a python module that provides testing functionality around the REST API in the Test Suite |
ASTERISK-21822: Adaptive CDR Not Written When Call Ends In Blind Transfer |
ASTERISK-21823: Segfault when autoload=yes in modules.conf |
ASTERISK-21824: SIP INVITES from Telphin broke in 1.8.21 |
ASTERISK-21825: [patch] websocket segmentation fault on certain invalid input |
ASTERISK-21826: [patch] Bad queue_log entry when removed member from queue via CLI |
ASTERISK-21827: [patch] Add kick all capability to app_confbridge's CLI command 'kick' |
ASTERISK-21828: [patch] app_meetme.so hints load as Unavailable instead of Idle on start up |
ASTERISK-21829: Bridge API Enhancements - finish connected line/redirecting handling in the bridging core |
ASTERISK-21831: Fix skipped cdr/blind-transfer-accountcode test for 12 |
ASTERISK-21833: New SIP Channel Driver - implement nominal registration tests - test 1 |
ASTERISK-21834: New SIP Channel Driver - implement nominal registration tests - tests 2 and 3 |
ASTERISK-21835: New SIP Channel Driver - implement nominal registration tests - test 4, 5, and 6 |
ASTERISK-21836: New SIP Channel Driver - implement off-nominal registration tests |
ASTERISK-21837: New SIP Channel Driver - implement nominal and off-nominal unregister tests |
ASTERISK-21843: Failed Dial() in a call file does not post a CDR record |
ASTERISK-21845: maxcalls exceeded, Asterisk sends out 480 and also BYE |
ASTERISK-21846: RINGNOANSWER event for an agent in queue, but data1 field is null |
ASTERISK-21847: Segfault due to dahdi_restart and round robin |
ASTERISK-21849: Transfer Asterisk queues are not seen in cdr reports |
ASTERISK-21854: Long Asterisk-version strings display improperly in the 'Connected to ...' line upon remote console connection |
ASTERISK-21855: Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available |
ASTERISK-21856: STUN never works when asterisk started without internet access |
ASTERISK-21857: Move Stasis-HTTP websocket URL from /ws to /stasis/events |
ASTERISK-21859: Confbridge doesn't tear down an empty conference bridge when all users were kicked via end_marked=yes. Also, side effect crashes. |
ASTERISK-21862: Add manager commands |
ASTERISK-21863: [patch] RTP Native Bridge Codec Change Handling - Appears to compare immediately after setting equal |
ASTERISK-21865: STRFTIME sometimes returns '0000-00-00 00:00:00' |
ASTERISK-21867: IAX Trunk Realtime - ipaddr gets set to (null) in iax_trunk table |
ASTERISK-21868: Asterisk REST API - Implement channel variables/global variables |
ASTERISK-21870: Asterisk REST API - Add dialplan location to the 'release back to dialplan command' |
ASTERISK-21872: high CPU usage ~15 seconds into call if rtpkeepalive set on channels when Asterisk is in a generic bridge and passing RFC2833 DTMF |
ASTERISK-21873: Asterisk API Improvements - filter channels that should never be shown |
ASTERISK-21875: Bridge API Enhancements - add CHANNEL(after-bridge-goto) feature |
ASTERISK-21876: Bridge API Enhancements - add CHANNEL(dtmf-features)=[tkhwx] feature |
ASTERISK-21877: Bridge API Enhancements - fix the Parking BUGBUG comments in trunk |
ASTERISK-21879: app_queue's autofill=yes effectively fails to deliver all calls when those calls are preceded by a call with a min/max penalty that can't be delivered |
ASTERISK-21882: Bridge API Enhancements - ensure that n-1 channels leaving a multi-party bridge ejects the last channel |
ASTERISK-21883: Asterisk API Improvements - refactor channel/bridge inspection commands to query the Stasis Cache |
ASTERISK-21884: Asterisk API Improvements - add AMI/CLI command for querying Stasis endpoints |
ASTERISK-21885: Asterisk REST API - modify JSON events to include an event type field; update swagger and code generation to use a discriminated union |
ASTERISK-21886: Bridge API Enhancements - add native bridging capabilities back to chan_dahdi |
ASTERISK-21892: Segfault after fax |
ASTERISK-21893: Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c |
ASTERISK-21894: [patch] Initial support for SIP/TLS tlsverifyclient |
ASTERISK-21895: Aborted in ast_hangup -> __ast_manager_event_multichan -> append_event |
ASTERISK-21896: Aborted in ast_bridge_call -> ast_cdr_dup_unique_swap -> ast_cdr_dup |
ASTERISK-21897: D option in Dial doesn't recognize "w" as a pause |
ASTERISK-21898: Read application does not set the variable |
ASTERISK-21899: Q850 Reason not forwarded between call legs |
ASTERISK-21900: help text syntax example for Read application delivered onto wiki incorrectly |
ASTERISK-21901: speex16 call to app_record with wav format results in a playable, but horrible sounding audio file |
ASTERISK-21902: Configuring asterisk to use a certificate generated by a peer |
ASTERISK-21903: [patch] Return proper result upon error when running some AGI commands |
ASTERISK-21904: Whisper problem when the channel between Agent and client is in hold |
ASTERISK-21905: Slow CODEC Translation compaired to the previous installation |
ASTERISK-21906: [patch] Fix memory leaks, invalid reads and more reported by valgrind |
ASTERISK-21907: Crash - segfault - When executing a MeetMeAdmin command that requires a member, without specifying a member |
ASTERISK-21908: Asterisk do not log source IP for Fake auth rejection |
ASTERISK-21911: Tearing down a registration throws a 403 back at the endpoint |
ASTERISK-21912: Call hang-up when issuing mixmonitor start |
ASTERISK-21913: Successive NOTIFY for MWI subscriptions isn't sent |
ASTERISK-21916: Call hangs when FILTER function is used in dial plan |
ASTERISK-21917: [patch] STUN crashes when SIP is bound to ipv4 and ipv6 |
ASTERISK-21918: WaitExten(5,m(default)) broken IAX2 channel |
ASTERISK-21919: Originate request cause an INVITE from "anonymous.invalid" domain |
ASTERISK-21920: IAX trunk timestamps set to zero when it's bridged SIP channel wraps RTP timestamps |
ASTERISK-21921: Verbose() ignores remote console VERBOSE message verbosity |
ASTERISK-21922: Add the ability to app_bridgwait to specify a particular bridge to place channels into |
ASTERISK-21923: Add the ability to app_bridgewait to specify various music and sound options |
ASTERISK-21924: Have the core bridging layer set the channel hang up cause on the channel/peer when the peer/channel breaks the bridge |
ASTERISK-21925: Clean up the parking API in res_parking |
ASTERISK-21930: [patch]WebRTC over WSS is not working. |
ASTERISK-21931: Make menuselect displays warning that Makefile has a modification time in the future |
ASTERISK-21932: [patch] ast_tls_cert: don't re-create generated files |
ASTERISK-21933: DTMF feature hook triggered for both caller and peer when peer initiates blind transfer |
ASTERISK-21938: PickupChan picks up answered call |
ASTERISK-21939: New SIP Channel Driver - add CLI/AMI commands that force actions |
ASTERISK-21941: menuselect possibly lies about speex dependencies |
ASTERISK-21943: Bridge API Enhancements - handle AgentLogin/AgentLogout in the Queue Log using Stasis |
ASTERISK-21944: Bridge API Enhancements - implement IAX2 native bridging (again) |
ASTERISK-21947: New SIP Channel Driver - use the proper bridging API function to get the bridged channel during direct media tests |
ASTERISK-21951: res_fax unsafely unlocks channel to perform an asynchronous goto to the 'fax' extension |
ASTERISK-21953: connectedline parameter not documented |
ASTERISK-21954: Local channel optimization needs to take into account frame hooks on the local channels. |
ASTERISK-21955: [patch] Asterisk responds 488 to session timers re-Invite in an active T.38 dialog after exactly 5 seconds |
ASTERISK-21956: MusicOnHold RealTime does not acknowledge announcement field |
ASTERISK-21959: SPAMMY = NOTICE[24695]: chan_sip.c:27899 handle_request_subscribe: Received SIP subscribe for peer without mailbox: |
ASTERISK-21960: ooh323 channels stuck |
ASTERISK-21963: CLONE - Callers on Queue are not being delivered to free agents |
ASTERISK-21964: SIP TLS Register statement fails if sip.conf register directive uses peer name. |
ASTERISK-21965: [patch] Bug-fixed version of safe_asterisk not installed over old version |
ASTERISK-21966: When FXS detects a fax, we should also disable callwaiting |
ASTERISK-21967: CFLAG Improvement to prevent compiler error in Virtual Machine environments |
ASTERISK-21968: Remove parkinglot from channel snapshots |
ASTERISK-21969: Odd events during Stasis origination |
ASTERISK-21970: Reconnects to an ARI websocket do not convey events for channels already in the application |
ASTERISK-21971: POST to /stasis/bridges/?type=[bridgetype] fails silently |
ASTERISK-21972: Bridge creation should allow for something more than mixing type during creation |
ASTERISK-21973: ARI /bridges/{}/addChannel should allow an optional parameter specifying a role |
ASTERISK-21974: ARI: Channels/bridges need MoH |
ASTERISK-21976: Set more than one codec in dialplan execution using SIP_CODEC (adapted chan_sip:try_suggested_codec) |
ASTERISK-21977: Stop potential message ordering issues between bridge and channel manager events |
ASTERISK-21978: Crash caused by RAII_VAR in test_json when loading module |
ASTERISK-21980: Error message for QUEUE_MEMBER when member is not in queue is unclear |
ASTERISK-21981: Pass-through support for Opus and VP8 formats |
ASTERISK-21991: [patch] - install a systemd service unit |
ASTERISK-21996: chan_iax2 fails to process network packets after a while |
ASTERISK-21997: [patch] - Incorrect Ring tone for Malaysia |
ASTERISK-21998: ChanSpy whisper mode doesn't completely mute one the channels |